24 Feb
2016
24 Feb
'16
3:16 p.m.
Hello Sebastien,
This won't answer your question, but I would rather use an sbc (
freeswitch, yate, asterisk, ...) to deal with the routing logic,
especially if you
want to leave some room for the future ( like transcoding, retrying a
call to a different carrier, ... ).
Kamailio can then be used as an awesome load-balancer/proxy in front of
your sbc farm.
That said, if you don't want/need to deal with the media, and just want
to work with the sig part, then you should look at the Uac module.
Best,
Tristan.
PS: The OPTION packets you saw are probably NAT keepalives that you can
deactivate.
On 02/24/2016 12:04 PM, Sébastien Brice wrote:
Hi everyone
I followed this guide
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
and got it working (101, 102 and 103) can call eachother.
But now i am trying to figure Asterisk's role out.
I am more an ipbx person and i am used to register providers trunk in asterisk/sip.conf
file, like this:
register =>
[peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension]
doing that i got plenty of OPTIONS request and 200 OK reply between my Kamailio and my
provider (and it is a bit noisy)
doing that i feel missing kamailio's logic and power to deal with externals trunk
provider
The thing is i need my authenticated users (101,102,103) be capable dialing my trunk and
requesting INVITE for non-local request.
What is the best way to achieve that?
My DID provider gave me user/passwd/realm.
I heard about avp special variables (auth_XXXX_avp and uac) and some snippets config that
could help me to go there.
Is that efficient to place the routing's logic to Kamailio and how to do that with my
ovh trunk?
thx you
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